As shown in FIG. 6, for example, as for an onboard audio system, since the distances DW R, DT R, DW L, and DT L (all of which are represented by solid lines in FIG. 6) from the two-way speaker units (right woofer WR, right tweeter TR, left woofer WL and left tweeter TL) placed in a vehicle to a listener differ, an acoustic image (represented by broken lines in FIG. 6) is pulled to the position of the closest speaker unit owing to Haas effect (precedence effect), and hence a good sound field cannot be obtained.
In this case, time alignment processing (time adjustment) is carried out so that sounds emitted from the four speaker units arrive at the listener position simultaneously by providing the individual signals with delay processing. Band-splitting time compensation signal processing devices have been known which correct, by adjusting the arrival time of the sounds in this way, the sound field bias resulting from the distance differences between the individual speaker units and the listening position of the listener.
Conventionally, to adjust relative time relationships between the speaker units having a plurality of bands split, a band-splitting time compensation signal processing device has been known, for example. It has, as shown in FIG. 7, independent amplifiers (power amplifiers 71 and 72) for individual speaker units prepared for each channel (right channel speaker units WR and TR, here), and has at their previous stage a band splitter 70 and delay circuits 73 and 74 for time axis adjustment. Here, only the R channel is shown, and for the L channel, the same circuit components are required additionally.
In this case, since the same number of amplifiers as the speaker units is necessary, problems arise of increasing the cost, complicating wiring and requiring a larger space.
On the other hand, there is a method that drives the speaker units, which have the bands split, with a single amplifier using a crossover network. In this case, a relative time difference between the band-split speaker units can be corrected by placing a digital signal processing circuit before the amplifier, by reproducing an impulse signal through the speakers, and by obtaining the inverse transfer function of the speaker system by observing the response waveforms followed by a convolution algorithm.
According to this method, however, the frequency characteristics and phase characteristics other than the time axis are corrected simultaneously, which means that it is impossible to adjust the time axis alone without changing the other characteristics. In addition, the values on the time axis cannot undergo fine adjustment independently.
In view of this, as for the method of driving the speaker units having a plurality of bands split with the single amplifier using the crossover network circuit, a listening position automatic compensation device is proposed that divides the band at about the same frequencies as the frequencies at which the crossover network circuit performs the band splitting before outputting to the amplifier, passes the individual signals after the division through delay circuits, and then mixes the signals again (see Patent Document 1, for example).    Patent Document 1: Japanese Patent Laid-Open No. 7-162985/1995
According to the technique disclosed in Patent Document 1, it corrects the bias in the sound field resulting from the distance differences between the individual speaker units and the listening position of the listener, and the disturbance of the frequency characteristics due to phase interference throughout the band.
However, it brings about a signal lost through the band-splitting circuit and a signal added double, which presents a problem of deteriorating the linearity of reproduced sounds, and produces a peak or dip in the frequency characteristics near the band-splitting frequency (crossover frequency) at the mixing.
The present invention is implemented to solve the foregoing problems. Therefore it is an object of the present invention to provide a band-splitting time compensation signal processing device capable of not only adjusting the time axis of each speaker unit independently, but also improving the linearity of the transfer characteristic at the listening position, and suppressing the occurrence of a peak or dip at the mixing point of time.